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<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=125203823-08022006>Look at the Manager command
'Redirect'</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=125203823-08022006></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=125203823-08022006>You can specif y two channels to be 'Redirect'ed to
each other.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=125203823-08022006></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=125203823-08022006>The two legs that do not have calls on them anymore
will drop.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=125203823-08022006></SPAN></FONT> </DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Wai
Wu<BR><B>Sent:</B> Wednesday, February 08, 2006 6:07 PM<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> RE:
[Asterisk-Users] Connecting to live calls<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=414270011-09022006><FONT face=Arial color=#0000ff
size=2>Sorry for being clear in the fast place. The scenario is like this. Two
calls coming into * and they are both on different extensions playing
messages. Some how a control program decides to bridge them together using the
Manager API. Is this possible without sending the calls to a meeting
room?</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]<B>On Behalf Of </B>Mark
Edwards<BR><B>Sent:</B> Wednesday, February 08, 2006 5:33 PM<BR><B>To:</B>
'Asterisk Users Mailing List - Non-Commercial Discussion'<BR><B>Subject:</B>
RE: [Asterisk-Users] Connecting to live calls<BR><BR></FONT></DIV>
<DIV class=Section1>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><!-- Converted from text/rtf format -->Wai.</SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">I think you need to
be a little clearer on exactly what you want to try and do. Can you please
outline a scenario for us. It’s not really clear what you mean by ‘two live
calls’. </SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">However, assuming
that each ‘live call’ has an A party and a B party, is this 4 parties you
want to connect up? If so, then I think the only way to do this is to use a
meeting room as you will effectively have a conference call between 4
parties.</SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Cheers,</SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Mark
Edwards</SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">www.switchnet.com.au</SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT> </P>
<P class=MsoNormal style="MARGIN-LEFT: 36pt"><FONT face=Tahoma size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">-----Original
Message-----<BR><B><SPAN style="FONT-WEIGHT: bold">From:</SPAN></B> Wai Wu
[mailto:wwu@Calltrol.com] <BR><B><SPAN
style="FONT-WEIGHT: bold">Sent:</SPAN></B> Thursday, 9 February 2006 8:58
AM<BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B>
asterisk-users@lists.digium.com<BR><B><SPAN
style="FONT-WEIGHT: bold">Subject:</SPAN></B> [Asterisk-Users] Connecting to
live calls</SPAN></FONT></P>
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size=3><SPAN style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P style="MARGIN-LEFT: 36pt"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Hi all,</SPAN></FONT> </P>
<P style="MARGIN-LEFT: 36pt"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Is there a way to connect two
live calls through the manager api without directing them to a meeting room?
Currently, I can connect them by sending them to a meeting room. However, I
don't know what the overhead is, and I kind of think that if I can connect
them or link them up, the overhead would be
minimum.</SPAN></FONT></P></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>